As some of you may already know, I did not make my first attempt at the exam although my score report indicated I was very close, let me be totally clear: It is very difficult, I personally found it MUCH harder than my routing and switching exam which frankly I cruised through, the exam is very long, so make sure you have good speed.
But I do not give up that easily! If Cisco thinks they have seen the last of me they are sorely mistaken, I have booked the lab again and am revisiting some areas of the blueprint that I know need work, the quick breakdown of areas I want to be 100 percent on are listed below:
- Switch QOS (I suck at this and I also _hate_ it, I find it very complicated)
- Gatekeeper Troubleshooting and SIP Call Flows/H323 Call Flows/MGCP Call Flow debugging
- I am going to investigate every possible way of doing IP to IP Gateways, supplementary services and get definitive answers on things like when MTP is needed, when it's not, what does work, what doesn't and more
- All the features in CUCM and CUCME including all the service parameters and other little bits that might affect them.
We all know about how to change our calling number when we make calls out to the PSTN right? It is fairly simple stuff, there is lot's of ways we can do it. For example on route-patterns, translation-patterns and transformation patterns in CUCM and on CME we can use voice translation-rules, dialplan pattern commands etc (although I personally now never ever use dialplan pattern.)
But what about the display on the phone when we DIAL a number, for example, When I dial 9911, how could I for example make this display as "911" or something else, and what order do these rules take affect?
Let's start from the top
H.323 and SIP interactions
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To summarize this right from the start: Basically any sort of digit manipulation you perform, be it a route-pattern, a transformation-pattern, a translation pattern or a voice translation-profile IS going to take affect and update the "To" display for callers. The only exceptions to this rule are dial-peer commands such as prefix or forward-digits (note that voice translation-rules DO affect the "to")
Here are the scenarios I performed and proved 100 percent in a lab environment at my home
First some quick lab information of my home setup:
1 Publisher, CUCM 7.0.3
1 Voice Gateway (Main Router) , Version 12.4(24)T2,
1 PSTN Gateway (Phone configured with "911" and "999")
1.
Prefix Command, Digit strip, forward-digits etc appear to have no affect on "To" display
tested by: configuring the following dial-peer
dial-peer voice 999 pots
destination-pattern 4444
forward-digits 0
prefix 911
!
Phone display showed "4444" despite the fact that 911 was being sent to the ISDN
True for both phones running straight off this CCME and phones connected via CUCM
Voice translation-rules:
Translation rules on either voice-ports or the dial-peer itself did affect the "To" and updated it. It also confirmed a suspicion of mine that the "prefix" and "Forward-digits" commands are applied AFTER the voice translation-rules are applied, even if the rule is configured on a voice-port itself. So the prefix and forward-digits are always the last digit manipulations to be done.
Testing methodology:
dial-peer with destination pattern 900
translation-rule converts 900 to 11 and is then applied to dial-peer
dial-peer has "prefix 9" configured
when calling 900, display updates to show "11" but 911 is sent to PSTN
CUCM elements:
Route-plan called party transformation masks and prefix's
- DO affect TO
Translation-pattern called party transformation masks and prefix's:
- DO affect TO
Transformation-patterns (called party)
- DO affect TO
this confirmed 100 percent with H.323 and SIP, I would love someone else to get the chance to test this in there own lab and confirm. If my testing instructions or methodology is not descriptive enough please let me know in the comments section and I will give more information!
Now we move ontop MGCP.
MGCP
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To summarize, MGCP "to" was affected for route-pattern called-party transform and translation-pattern called party transforms but NOT updated if a transformation-pattern was applied, voice translation-rules on the voice-port did not work as this is MGCP so they would not take effect. Below is the testing methodology.
Route-pattern created for "11" and called-party digit manipulation said to prefix 9
when ringing "11" phone "to" displayed "911"
Translation-pattern created for "4567" and called-party digit manipulation set to "11"
this then matched the route-pattern shown above
Phone displayed "911"
Transformation-pattern created to match "11" and prefix a 9 infront of it, applied to gateway
route-pattern changed to not perform any digit manipulation and simply send "11" to gateway
transformation-pattern then matched this, so call went out to PSTN as 911 but "TO" display showed as "11"
I hope this helps someone out there!
Kind Regards
Peter
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